Perceptually Enhanced Bit-Plane Coding for Scalable Audio

2003

This papers presents an embedded fine grain scalable perceptual and lossless audio coding scheme. The enabling technology for this combined perceptual and lossless audio coding approach is the Integer Modified Discrete Cosine Transform (IntMDCT), which is an integer approximation of the MDCT based on the lifting scheme. It maintains the perfect reconstruction property and therefore enables efficient lossless coding in the frequency domain. The close approximation of the MDCT also allows to build a perceptual coding scheme based on the IntMDCT. In this paper a bitsliced arithmetic coding technique is applied to the IntMDCT values. Together with the encoded shape of the masking threshold a perceptually hierarchical bitstream is obtained, containing several stages of perceptual quality and extending to lossless operation when transmitted completely. A concept of encoding subslices is presented in order to obtain a fine adaptation to the masking threshold especially in the range of perceptually transparent quality.

2004 IEEE International Conference on Acoustics, Speech, and Signal Processing, 2004

Lossless coding will become the latest extension of the MPEG-4 audio standard. In response to a call for proposals, many companies have submitted lossless audio codecs for evaluation. The codec of the Technical University of Berlin was chosen as reference model for MPEG-4 Audio Lossless Coding (ALS), attaining working draft status in July 2003. The encoder is based on linear prediction, which enables high compression even with moderate complexity, while the corresponding decoder is straightforward. The paper describes the basic elements of the codec, points out envisaged applications, and gives an outline of the standardization process.

2007

In this paper we present Fast Perceptual Quantization (FPQ), a novel procedure to quantize and code audio signals. It employs the same psychoacoustics principles used in the popular MPEG/Audio coders, but substantially simplifies the complexity and computational needs of the encoding process. FPQ is based on defining a hierarchy of privileged quantization values so that the masking threshold calculated through a psychoacoustic model is leveraged to quantize the real values to the privileged ones when possible. The computational cost of this process is very low compared to MP3’s or AAC’s quantization/coding loops. Experimental results show that it is possible to achieve nearly transparent coding using as few as approximately 100 quantization values. This leads to very efficient bit compaction using Huffman or arithmetic coding so that nearly state-of-the-art performance can be achieved in terms of quality/bit-rate trade-off. Since quantization and codification (bit compaction) proced...

2004

Lossless coding will become the latest extension of the MPEG-4 audio standard. The lossless audio codec of the Technical University of Berlin was chosen as reference model for MPEG-4 Audio Lossless Coding (ALS). The MPEG-4 ALS encoder is based on linear prediction, which enables high compression even with moderate complexity, while the corresponding decoder is straightforward. The paper describes the

2004

Embedded lossless audio coding attempts to combine the higher compression ratios of perceptual coding with the perfect reconstruction of the original signal provided by lossless coding. This paper examines the residual signal of a perceptual audio coding base layer and considers its usage as an embedded bitstream for an embedded lossless coder. It is shown that the residual signal of a lossy perceptual audio coder retains correlation that can be exploited in lossless compression. This allows an embedded stream to be provided in a lossless coder with approximately 6% overhead over pure lossless audio coding. A 6% overhead appears to be a minimal cost for the backward compatibility and scalability afforded by embedded streams.

2016

Scalable to lossless audio compression based on perceptual set partitioning in hierarchical trees (PSPIHT)

IEEE Transactions on Speech and Audio Processing, 1995

ABSTRACT A new audio transform coding technique is proposed that reduces the bitrate requirements of the perceptual transform audio coders by utilizing the stationarity characteristics of the audio signals. The method detects the frames that have significant audible content and codes them in a way similar to conventional perceptual transform coders. However, when successive data frames are found to be similar to those sections, then their audible differences only are coded. An error analysis for the proposed method is presented and results from tests on different types of audio material are listed, indicating that an average of 30% in compression gain (over the conventional perceptual audio coders bitrate) can be achieved, with a small deterioration in the audio quality of the coded signal. The proposed method has the advantage of easy adaptation within the perceptual transform coders architecture and adds only a small computational overhead to these systems

2003 IEEE International Conference on Acoustics, Speech, and Signal Processing, 2003. Proceedings. (ICASSP '03)., 2000

For the purpose of efficient audio coding at low rates, a new bit allocation strategy is proposed in this paper. The basic idea behind this approach is "Give bits to the band with the maximum NMR-Gain/bit" or "Retrieve bits from the band with the maximum bits/NMR-Loss". The notion of "bit-use efficiency" is suggested and it can be employed to construct a bit assignment algorithm operated at band-level as compared to the traditional framelevel bit assignment methods. Based on this strategy a new bit assignment scheme, called Max-BNLR, is designed for the MPEG-4 AAC. Simulation results show that the performance of the Max-BNLR scheme is significantly better than that of the MPEG-4 AAC Verification Model (VM) and is close to that of TB-ANMR [3], which is the (nearly) optimal solution. Moreover, the Max-BNLR scheme has the advantages of low computational complexity comparing to TB-ANMR.

Speech and Audio …, 2002

Abstract—This paper proposes a versatile perceptual audio coding method that achieves high compression ratios and is capable of low encoding/decoding delay. It accommodates a variety of source signals (including both music and speech) with different sampling rates. It is ...

[1991] Proceedings, Advanced Computer Technology, Reliable Systems and Applications, 1991

Based on frequency-domain techniques, coding of high-quality audio with bit rates down to 64 kbit/s is possible. This performance is achieved using perceptual coding. Transform coding can be used to get the best performance at very low bit rates. Real-time implementations of several types of low bit rate codecs have been developed. Standardization of low bit rate audio coding systems

2009 IEEE International Conference on Acoustics, Speech and Signal Processing, 2009

This paper proposes a new bit plane coding method for signed integer sequences. This method consists in mapping successive bit planes onto quinary symbols (+, -, 0, 1, EoP ), where the symbol "EoP " stands for "End of Plane", and applying arithmetic coding. Sign bits are efficiently coded in combination with the corresponding most significant bit of non-zero integers. Moreover, bit planes are scanned and coded in a non-sequential manner to exploit the correlation between successive planes. Results for conversational transform coding of wideband speech and audio signals -sampled at 16 kHz -show that the performance/complexity of the proposed bitplane coder is near equivalent to non-embedded coding (stack-run coding), while offering additional flexibility (bitstream scalability).

2003

This paper proposes a technique for scalable and also provides for lossless compression. It reduces smooth objective scalability, in terms of SegSNR, from lossy to lossless compression. The proposal is built around the perceptual SPIHT algorithm, which is a modification of the SPIHT algorithm and is introduced in this paper. Both objective and subjective results are reported and demonstrate that the proposed method performs comparably with the MPEG-4 AAC coder at 16, 32 and 64 kbps, yet also achieves a scalable-to-lossless architecture.

Proceedings of the 5th WSEAS international …, 2005

Abstract: -- In this paper we present an efficient high quality perceptual audio coding scheme with a novel dynamically switched filter bank design and an associated bit allocation strategy. An additional processing block is added prior to the decomposition subband ...

2017 IEEE 6th Global Conference on Consumer Electronics (GCCE)

This paper describes experimental evaluations of encoding parameters that are appropriate for MPEG-4 Audio Lossless Coding (ALS) to compress high-resolution audio. MPEG-4 ALS Simple Profile defines the values of encoding parameters, such as the maximum sampling frequency and quantization bit depth, for making it easier to implement in the receiving applications. However, ALS Simple Profile does not define values for 96-kHz high-resolution audio. Therefore, we propose a range of values for 96-kHz high-resolution audio by extending the ALS Simple profile and evaluating its performance.

Digital Signal Processing, 2003

This paper presents a very low-complexity audio codec that provides audio playback quality similar to the MPEG-I/audio level 3 codec at 64 Kbps for a monophonic-channel signal. This welldesigned low-sophistication scheme uses a simple noise-masking model, a specialized nonuniform quantizer, an effectively recursive refinement module, and an adaptive arithmetic coder with multiplication-free adaptation. For bit allocation, we propose an appropriate nonuniform quantizer incorporating noise-masking effects and designed for fast implementation as well as efficient acceleration of the proposed refinement process. This recursive refinement algorithm effectively improves recovered perceptual audio quality after quantization. The adaptive arithmetic coder uses two fast adaptation algorithms that do not require multiplication to quickly obtain efficient bit allocation. A Taylor series expansion is used to simplify the frequently executed functions in the masking threshold formulas. Thus, the proposed high-quality audio codec appears to be a very valuable consumer electronic approach or a software solution at low cost.